Method and system for wireless real-time collection of multichannel digital audio

ABSTRACT

In this application is described a method and a system for collecting streaming multi channel digital isochronous data from multiple independent digital signal sources. The method is used for collecting streaming multi channel digital isochronous data, e.g. audio data, in a standard wireless local area network transmission system where bandwidth is reserved for both contention-based traffic and contention free traffic and the audio data ( 10 ) formed by samples ( 9 ) is organized in audio frames ( 174 ) and sent to receivers ( 6 ) using multicasting, within consecutive beacon intervals ( 137 ). In accordance with the invention the contention free traffic ( 138 ) of the beacon interval ( 137 ) is adjusted to an optimum value, and the length of the beacon interval ( 137 ) is adjusted such that a required amount of audio data ( 9 ) can be sent to the receivers ( 6 ) with minimum system delay.

The invention relates to a method according to the preamble of claim 1for wireless real-time signal collection from several independentsources for mainly audio purposes.

The invention relates also to a system according to the preamble ofclaim 6 for wireless signal collection from several independent sourcesfor mainly audio purposes.

The invention relates to an error control method and system and asynchronization method and system for the said purposes.

The object of this invention is typically a system with the associatedapparatus and method for the isochronous, electromagnetic disturbanceresistant, wireless transfer of highest studio-quality multi-channeldigital audio signals from several independent but synchronized sourcesto a central station. This same method can also be used as the basis ofthe high-speed transmission of other digital information with the samekind of real-time and bandwidth requirements such as synchronizeddigital measurements from several independent sources.

INTRODUCTION

With the currently known technique, the studio-quality multi-channeldigital audio signal from a set of independent signal sources such asmicrophones is first transferred to the multi-channel digital mixer withthe balanced per-channel electrical cables. The analog-to-digitalconversion is performed in the mixer and the channels are finallyrecorded to a digital storage device after the required balancing andmixing operations have been applied. Also, a transmission method withspecial purpose radio links is known. The physical analog transmissionpath injects several degrading effect such as noise, interference,distortion, group delays, amplitude and phase errors to the quality ofthe original signal. The cabling is often clumsy and can be messylooking especially in concert occasions. With careful design andbalancing of cables and their wiring layout, these effects can belimited to some extent but seldom completely eliminated. The number andbulkiness of the cables, the need for careful design and tediousinstallation work increase the costs as well as required skills andtime. Cables and their electromechanical connectors are also prone tomechanical failures, which are hard to find and fix. These problems areespecially harmful in public performances when the performers and ofteneven the audience move among the cables. Under these conditions, therecan be a real hazard of harm and injury with the cabling. Duringartistic tours, the audio equipment is installed and uninstalledfrequently to and from varying environments, which multiplies theseproblems, efforts, and costs.

With the use of modern capacitive microphones, having integrated andoptimised preamplifiers within them, the analog signals can be of lowerpower level and also the more noise and interference resistantdifferential signalling can be employed. The generation of multi-channeldifferential signals requires, however, rather expensive high-qualityanalog electronics plus costly differential cabling and connectorsindependently of what type of microphones are used.

The currently available wireless audio microphone systems arenon-standard radio or infrared solutions typically using lossy audiocompression methods thus resulting compromised performance. They aretherefore mainly used for supportive purposes such as public addressvoice transmission.

The aim of this invention is to solve problems relating to theisochronous real-time collection of the highest studio-quality streamingdigital audio signals associated with the techniques described above byconstructing a novel, international standards compliant wireless localarea network (WLAN) based data communication system, transmitters,receiver plus the necessary firmware and software for the efficientrestricted area collection of digital audio signals and the testing,configuration, management and control of such systems.

The invention is based on the idea that the digital information istransferred using special speeded-up sequential unicast from thedifferent transmitter stations to the central collecting station in thestudio-quality digital format with electro-magnetic radio waves withoutdedicated signal cables using typically internationally standardized andhigh-volume produced wireless local area networking (WLAN) components.The analog signal is converted to the digital form directly at thesignal source and fed locally to the associated WLAN transmitter. Thisguarantees the ultimate sound quality at the microphone transmitter.Because of the application of the mass-produced WLAN technique and itscommercial components plus the very small number of additional standardintegrated circuits, the cost of the development work and the actualsystem components can be kept very reasonable. This part of the systemis typically powered by a rechargeable battery pack, which additionallyhelps in achieving noise free source signals.

The method introduced here replaces the wired lines with the standardcommercial wireless local area network technology as specified in theIEEE 802.11 series of standards. The special characteristics requiredfor the uncompressed real-time transfer of multi-channel studio-qualityaudio signals have been implemented by the innovative choice of WLANsystem coordination functions, communication modes, and controlparameters together with a special upper layer firmware implementing thespeeded-up sequential unicast.

In accordance with a preferred embodiment of the invention the audiodata formed by samples is organized in audio frames and sent from theindividual microphone stations to the receiver station withinconsecutive beacon intervals, using coordinated, speeded-up unicastmessaging. According to the WLAN standards, two co-existing transmissionservices are possible. The usual mode, widely used in commercial datacommunication products, is called the contention-based service. Theother mode, used seldom, but accurately specified in the IEEE 802.11standard, is called the contention-free service, and it is the basis forthis invention. Special beacon frames are used to control the switchingbetween these two modes of operation. The length of the beacon intervalis a programmable parameter and it is adjusted with this invention sothat an optimum amount of isochronous audio signal data can be sent tothe receiver, with a minimum of system delay. This optimal amount is inone preferred embodiment of the invention for the required amount ofisochronous audio signal data for high quality audio broadcasting andrecording.

In accordance with another preferred embodiment of the invention, anerror control system optimised for isochronous digital audio transfereither minimizing the need or totally eliminating the need forretransmissions is used, where the received signal is correlated withother channels, is used for error correction purposes.

In accordance with a third preferred embodiment of the invention, thetransmitters and their signal sampling are synchronized in a coordinatedunicast system with the help of an end-of-frame interrupt, generated bythe control frame terminating each beacon interval, at the exactly sameinstance within each beacon interval. This synchronization is furtherutilized to trigger the accurate coherent sampling of the audio signalsof the independent sources and as the reference instance for theindividual timers of the signal source transmitter timers that triggerthe coordinated unicast transmission at the proper instance so that eachtransmitter is active at the right period of time without interferingwith others.

In accordance with a fourth preferred embodiment of the invention, thetransmission order and sequential timing of the transmitters aresynchronized in a coordinated speeded-up burst unicast system with thehelp of an end-of-frame interrupt, generated by the control frameterminating each beacon interval, at the exactly same instance withineach beacon interval and accurate timers in transmitters triggering theactual frame transmission at the right instance of time. This speeded-uparrangement guarantees the best possible usage of the WLAN bandwidthfrom a set of independent transmitters to a single receiver.

More specifically, the method according to the invention ischaracterized by what is stated in the characterizing part of claim 1.

Further, the system according to the invention is characterized by whatis stated in the characterizing part of claim 6.

With the help of the invention significant benefits may be obtained.

With the coordinated per-signal-source transmission of thestudio-quality digital audio, all the error factors associated to thetraditional signal path can be avoided. Performing the digital-to-analogconversion immediately at the signal source itself maximizes the soundquality by localizing the propagation path of the analog signal on thefixed and optimized converter circuitry in accordance with oneembodiment of the invention.

The signal cables, their connectors and differential signaltransmitter/receivers and related material and installation work can becompletely avoided. This eliminates all the cost, failure, andinstallation problems associated with them. As mass produced standardWLAN technique is the basis of the invention, its production cost can bemade very low in accordance with one embodiment of the invention.

As the coordinated, speeded-up burst unicast transmission mode andfrequent multicast synchronization are utilized, the samplingsynchronization and the inter-channel phase errors can be effectivelyeliminated in accordance with one embodiment of the invention.

As optimized transmission frame sizes are used, the system level delayas well as the buffering requirements can be minimized to aninsignificant level in accordance with one embodiment of the invention.The proper varying of the frame size further guarantees the smooth, evenflow of the data stream.

With the help of the error control method in accordance with oneembodiment of the invention a simple and fast best-effort audio errorcorrection scheme can be obtained.

In the following, the invention will be described in more detail withreference to the exemplifying embodiments illustrated in the attacheddrawings in which

FIG. 1 shows as a block diagram a general system configuration of theinvention.

FIG. 2 shows as a block diagram an example transmitter station inaccordance with the invention.

FIG. 3 shows as a block diagram another example transmitter station inaccordance with the invention.

FIG. 4 shows as a block diagram an example receiver in accordance withthe invention.

FIG. 5 shows the audio data structure representing one multi channelaudio sample in accordance with the invention.

FIG. 6 shows a data structure representing one audio sample 16-tuplewith the appended error control blocks in accordance with the invention.

FIG. 7 shows with the help of the data structure of FIG. 6, the errorcorrection principle in accordance with the invention.

FIG. 8 shows as a block diagram the Medium Access Control (MAC)architecture, which can be used with the invention.

FIG. 9 shows as a data structure the general MAC frame structure, whichcan be used with the invention.

FIG. 10 shows as a data structure the WLAN frame control field, whichcan be used with the invention.

FIG. 11 shows as a block diagram the possible medium access control(MAC) addresses, the multicast version of which can be used with theinvention.

FIG. 12 shows as a data structure the generic beacon frame, which can beused with the invention.

FIG. 13 shows as a data structure a beacon frame in accordance with theinvention.

FIG. 14 shows as a data structure a capability information field, whichcan be used with the invention.

FIG. 15 shows as a data structure information elements, which can beused with the invention.

FIG. 16 shows as a data structure the Traffic Indication Map (TIM)element format, which can be used with the invention.

FIG. 17 shows as a data structure the Extended Rate PHY (ERP)information element, which can be used with the invention.

FIG. 18 shows as a data structure an extended supported rates element,which can be used with the invention.

FIG. 19 shows as a data structure the Contention-Free (CF) parameter setelement, which can be used with the invention.

FIG. 20 shows as a data structure a CF-End Frame, which can be used withthe invention.

FIG. 21 shows as a data structure an ERP-OFDM PHY frame structure, whichcan be used with the invention.

FIG. 22 shows as a graph the bandwidth requirement for the invention.

FIG. 22 a shows a detail of FIG. 22.

FIG. 22 b shows a detail of FIG. 22 a.

FIG. 23 shows as a table the number of 16×24-bit sample records inconsecutive data blocks in accordance with the invention, relating toproper sequencing of digital audio for transmission.

FIG. 23 b shows as a table the number of 24-Bit samples for 250transmission cycles of the 16 individual signal sources.

FIG. 24 shows as a graph the jitter behaviour in accordance with theinvention.

FIG. 24 a shows as an enlarged graph the jitter behaviour in accordancewith the invention and FIG. 24.

FIG. 25 shows as a block diagram a general data structure in accordancewith the invention relating to the worst-case transmission timing.

FIG. 25 a shows as a table the timing of the beacon signal.

FIG. 25 b shows as a graph the transmission durations of the invention.

FIG. 26 shows as a flow chart audio input processing in accordance withthe invention.

In this document, the following terms will be used in connection withthe inventions.

-   1 WLAN, Wireless Local Area Network-   2 mixer or recorder-   3 collector receiver-   4 Base station-   5 Remote controller-   6 audio source-   7 transmitter-   8 signal source subsystem-   9 audio data format/sample-   10 transmission level audio data format-   11 error correction code-   12 MAC sublayer, MAC=Medium Access Control-   13 Distributed coordination function-   14 Point coordination function-   15 Contention-free communication services-   16 Contention-based communication services-   17 General MAC frame structure-   18 Frame control-   19 Duration/ID-   20 Address 1-   21 Address 2-   22 Address 3-   23 Sequence Control-   24 Address 4-   25 Frame body-   26 FCS, Frame Control Sequence-   27 MAC Header-   28 MAC Frame-   29 WLAN frame control field-   30 Protocol version-   31 type-   32 Subtype-   33 To DS, DS=Distribution System-   34 From DS-   35 More Frag-   36 Retry-   37 More data-   38 Pwr Mgt-   39 WEP, Wired Equivalent Privacy-   40 Order-   41 Individual Address-   42 Group Address-   43 Unicast Address-   44 Multicast address-   45 Broadcast address-   46 Generic Beacon Frame-   47 Frame control-   48 Duration-   49 Destination address-   50 Source address-   51 BSS ID-   52 Sequence control-   53 Frame body-   54 FCS-   55 Time stamp-   56 Beacon interval-   57 Capability info-   58 SSID, Service Set IDentity-   59 Optional fields-   60 Beacon frame as used in this invention-   61 Frame control-   62 Duration-   63 Destination address-   64 Source address-   65 BSSID, Basic Service Set IDentity-   66 Sequence control-   67 Frame body-   68 FCS-   69 Time stamp-   70 Beacon interval-   71 Capability info-   72 SSID-   73 CF parameter set-   74 TIM, Traffic Indication Map-   75 ERP, Extended Rate PHY-   76 Extended rates-   77 Element format-   78 Element ID-   79 Length-   80 Information-   81 TIM element-   82 Element ID-   83 Length-   84 DTIM Count, DTIM=Delivery Traffic Indication Map-   85 DTIM Period-   86 Bitmap Control-   87 Partial Virtual Bitmap-   89 ERP information element-   90 Element ID-   91 Length-   92 Non ERP-present-   93 Use protection-   94 Barker Preamble mode-   95 r3-r7-   96 Extended Supported Rates element format-   97 Element ID-   98 Length-   99 Extended Supported rates-   100 CF Parameter Set element format, CF=Contention-Free-   101 Element ID-   102 Length-   103 CFP Count-   104 CFP, Contention-Free Period-   105 CFP Max Duration-   106 CFP DurRemaining-   107 CF-End frame-   108 MAC-header-   109 CF-end MAC Frame-   110 frame control-   111 Duration-   112 RA, Receiver Address-   113 BSSID-   114 FCS-   115 ERP-OFDM PHY Frame structure, OFDM=Orthogonal Frequency Division    Multiplexing-   116 Coded/OFDM-   117 PSDU, Protocol Service Data Unit-   118 PLCP Preamble-   119 SIGNAL-   120 Rate-   121 Reserved-   122 LENGTH-   123 Tail-   124 Parity-   125 Service-   127 Frame control-   128 Duratrion/ID-   129 Address 1-   130 Address 2-   131 Address 3-   132 Sequence Control-   133 Address 4-   134 Frame body-   135 FCS-   136 Area of interest-   137 WLAN Repetation period/Beacon interval (N*TU)-   138 Foreshortened contention-free Period-   139 Multiplexer and receiver-   140 Serial to parallel converter-   141 Buffer-   142 USB host controller, USB=Universal Serial Bus-   143 USB inputs-   144 S/PDIF-inputs, S/PDIF=Sony/Philips Digital InterFace-   145 Analog inputs-   146 Analog buffers and multiplexers-   147 A/D-converters, A/D=Analogue-to-Digital-   148 MAC/baseband Processor-   149 Microcontroller-   150 D/A-converter and filter, D/A=Digital-to-Analogue-   151 Select analog input-   152 A/D-conversion-   153 Select Digital input-   154 24-bit reformatting-   155 Select audio input-   156 Number of channels 8-   157 No-   158 Yes-   159 generate FEC and write to buffer, FEC=Forward Error Correction-   160 Calculate missing channels-   161 generate FEC and write to buffer-   162 sample i−1-   163 sample i-   164 sample i+1-   165 corrected sample i-   166 ESS, Extended Service Set-   167 IBSS-   168 CF Pollable-   169 CF Poll Request-   170 Privacy-   171 Reserved-   172 Antenna-   173 Most significant bits-   174 Audio MAC frame-   175 Control MAC frame

System

In accordance with FIG. 1, the system comprises one or several audiosignal sources 6, which may be either digital or an analog sources. InFIG. 1 these are represented by studio microphones. The sources 6 aredigitised, if necessary, and fed to the WLAN adapter and transmitter 7,which includes an antenna arrangement for robust wireless transmissionto the collector receiver 3 and from there to the sound consoles,mixers, recorder(s) 2 or to broadcast subsystems. The receiver 3 and thebase station 4 are typically controlled by a remote controller 5 or acomputer. The signal from the signal station 7 is sent via a WLAN basednetwork 1 using a sequence of isochronous, coordinated unicast messagesto the receivers 4 from the signal source subsystem 8, consisting, forexample, of several microphones 6. In other words the audio signal fromsources 6 is transformed into digital data by elements 7 and transferredto the collector receiver as standard WLAN digital data.

Transmitter Base Station

FIG. 2 shows a simple example version of the collector receiver basestation 4 and the audio storage and broadcasting equipment 2. Thecollector receiver base station 4 is typically a 108 Mbit/s extendedIEEE 802.11g WLAN MIMO Access Point station, which receives a specifiednumber of digital audio signals from the source transmitters. 108 Mbit/sis practically the lowest possible standard bit rate for the system ofthis invention. In the future, higher WLAN transmission speeds areexpected and can be used to increase the number of signal sources inproportion to the increased transmission speed. They will also make itpossible to improve the error correction methods using selectiveretransmissions. The received digital analog signals from the sourcetransmitters it is converted to S/PIDIF or AES3 bit streams forprocessing, recording, and broadcasting.

Within the collector receiver station 3 there is a 48 KB memory ringbuffer 141 or FIFO buffer for the intermediate storing of the incomingdata. After initialization the collector receiver station 3 uses thecontention mode traffic to initialise the signal sources. Each source isidentified based on its unique MAC address and is assigned a sequencenumber ranging from 1 up to a maximum of 16. This sequence number isused as the basis of the coordinated sequential speeded-up unicasttransmission described later. To start the collection the collectorstation changes its operation to the contention-free mode setting thebeacon interval to 6 TUs and sending to the source stations a command tostart the signal sampling from the synchronizing end-of-frame interruptof the next CF-End control frame. From this point the coordination ofthe transmission is allocated to the cooperating signal source stationsas described later. The WLAN part of the collector receiver station (andthe source transmitters) conforms to the IEEE 802.11g standard with therange and transmission rate extensions introduced by Atheros Inc. andAirgo Inc. A MIMO antenna arrangement 172 is typically also used. Thenominal bit rate is 108 Mbit/s. These implementations of the extendedIEEE 802.11g WLANs also contain a powerful transmission error correctionmechanism that effectively distributes the eventual transmission pathburst errors to single bit reception errors at reception and is capableof correcting all of them on the octet level. This feature is takenadvantage of in the specified application layer forward error correctionmethod.

Contention-based, individually addressed messaging between the basestation 4 and the receiver stations is used for the configuration,status monitoring, and control of the signal transmitters as well as thesignal source equipment attached to them. There is an infrared handheldremote controller receiver, a USB 2.0 computer communicationreceiver/transmitter and a USB 2.0 general-purpose receiver/transmitterfor Bluetooth and WLAN handheld remote controller adapters in thecollector receiver station 3.

System Configuration, Monitoring, and Control

The system configuration, monitoring and control are done from thehandheld remote controller(s) or from a (personal) computerapplication(s) as described above.

Source Transmitters

According to FIG. 4 the receiver 6 typically consists of a MIMO antennasubsystem 172, the IEEE 802g conformant WLAN circuit with the Atheros orAirgo range and transfer rate extensions. There are typically softwarecontrolled multi-color LEDs to aid the recognition and status of theindividual signal sources 7 for the configuration, status monitoring andcontrol operations. The WLAN is operated at the nominal speed of 108Mbit/s. The received audio data stream is buffered into a 48 KB inputring or FIFO memory buffer and the source signal transmission from thebuffer is started using a hardware timer controlled by the CF-Endend-of-frame interrupts and the driver firmware. The data of thedifferent sources is combined by a 32-bit processor 149 and fed to aS/PIDIF and AES3 parallel-to-serial converter 150 followed by opticaland coaxial cable driver electronics and corresponding connectors. Theoutput channel mode selection is done by the configuration and controlsoftware over the contention communication service of the WLAN.

The source transmitters 6 of the up to 16 channels each have an internalcrystal-derived clock to generate the 192,000 Samples/s clock. Theseclocks are restarted by the end-of-frame interrupt generated by theCF-End control message of each of the 6,144 μs transmission slot to keepthe independent signal sources and their sampling operations accuratelymutually synchronized.

Remote Control Terminals

Two methods exist for the control of the system, a battery-poweredhandheld control terminal 5 and a software application available forseveral platforms including Linux, MS Windows, Apple, and Symbianoperating systems.

Handheld Remote Controller

The handheld remote controller 5 contains a keypad, a small display, aprocessor and a communication link to the base station. The keypadfunctions allow the selection of the output ports 2, the signal sourcegroup 8 and individual signal source 7 configuration and control. Signalsource groups 8 as well as individual sources 7 can be smoothlyactivated and deactivated and their programmable features can beremotely adjusted. The handheld remote controller communicates with thecollector receiver station 4 via an infrared, Bluetooth or WLAN link.The receiver station 4 relays the controls to signal sources through theindividual signal transmitters using contention mode communication andeither group or individual addressing. There is a panic key and functionin the remote controller 5 that causes the smooth immediate muting ofall signal sources 7.

Remote Control Software

The system described above can be fully controlled by a computer runningthe configuration, monitoring, and control application software. Thecommands and responses are communicated with the transmitter basestation using a Bluetooth, IrDA, LAN, WLAN, or USB link.

Method

According to FIGS. 5 and 6 the invented apparatus transmitsisochronously, in real time, up to 16 fully independent butsynchronized, strongly encrypted and uncompressed channels of 24-bit 192000 Sample/s digital audio streams 11 from the individual signal sourcesto a common collector receiver station. A group 10 of 688 (orexceptional 689) discrete 24-bit samples 11, totalling 2 064 (or 2 067)sample octets, will be called transmission level source data blockformat in the rest of this presentation. The sustained application leveldigital audio data bandwidth requirement is thus 73,728 Mbit/s.Additionally there are the overheads caused by the PHY and MAC framing,encapsulation with the Advanced Encryption Standard based CCMPencryption, and the effects of the IEEE 802.11 contention traffic timeallocation. These make even the largest IEEE 802.11g WLAN bit rate of 54Mbit/s insufficient for this application. With today's standard WLANtechniques, the required performance cannot be achieved. The noveltransmission method described below is based on the innovative use ofthe contention-free speeded-up unicast transmission with the PointCoordination Function (PCF) as specified in the IEEE 802.11 standards.With careful parameter tuning the bandwidth of the WLAN can be optimallydivided between the PCF contention-free medium access mode and the usualDecentralized Control Function (DCF) contention access mode so that theisochronous multi-channel digital audio transfer becomes possible. Withthe 108 Mbit/s extension of the IEEE 802.11g WLAN network and by usingthe ERP-OFDM PHY layer framing it is possible to transmit the aimedsixteen (16) independent 24-bit, 192 kSample/s digital audio streamsisochronously together with normal contention based WLAN data traffic.The same is, of course, also possible with the highest bit rates of theIEEE 802.11 n equal or higher than 108 Mbit/s. The high number ofchannels, the high resolution, and the high sampling rate guarantee thewireless collection of the best sound quality commercially availabletoday.

Data Structure

According to FIG. 25 the aim of the invention is to transfer enoughaudio blocks (transmission level audio data format) 10 in order tocollect high quality audio sound. Firstly, the beacon interval 137defined by the software settings has to be chosen correctly in order toachieve the aim. The beacon signal, defining the length of the beaconinterval 137, is sent in intervals defined by an integer in the IEEE802.11g WLAN standard. The value of this integer may have values from 1to N. In other words, beacon interval 137 is a product of the beaconinteger and time unit (TU). The length of one TU in IEEE 802.11g WLANstandard is 1,024 μs and therefore the beacon interval 137 is a multipleof TUs (1,024 μs). However, the standard defines, that in each beaconinterval 137 there should be enough time reserved for the contentiontraffic, more precisely enough time for a maximum size frame, ACK, 2slot times and 2 SIFS. In accordance with the invention, an optimumvalue for the number of time units TU for a beacon interval 137 is foundto be 7. The optimum amount can be defined also as a sufficient amountin one preferred embodiment of the invention. This gives enough time tosend 32 audio MAC frames 174 within one beacon interval 137. Each audioMAC frame 174 includes 688 or 689 transmission level audio data formatblocks 10, the number of these blocks is defined in accordance with thetable of FIG. 23. In this figure one row represents the content of theaudio MAC frames 174 in one contention free period 138 of a beaconinterval 137. As can be seen from FIG. 23, a predetermined sequence isrepeated after each 125 beacon intervals. With the help of this detailedsequence, the average flow rates of the audio sources and WLAN outputare matched, and the jitter can be held at the minimum, as shown in FIG.24. This also results in a minimum requirement of buffer memory both inthe transmitter and in the receivers 6.

Bandwidth Division

According to FIG. 25, in order to guarantee the timely transport ofaudio source data, the highest possible repetition rate ofcontention-free periods 138 must be realized. At the same time, themaximum fraction of the network capacity must be reserved for the audiotraffic. The IEEE 802.11 standard requires that there must be enoughcontention traffic time within each repeating contention-free intervalfor the transmission of one maximum size data frame together with itsacknowledgement frame plus two SIFS periods and two slot times. With the108 Mbit/s bit rate and with the ERP-OFDM PHY framing this requirementequals to 212+40+2×10+2×9=290 μs. As described in the IEEE 802.11standard, the contention traffic in the beginning if the contention freeperiod 138 may foreshorten the contention period by a maximum value ofthe sum of an RTS control frame, a CTS control frame, one maximum sizedata frame, an ACK control frame plus four SIFS. With the 108 Mbit/s bitrate and with the ERP-OFDM PHY framing this requirement is equal to40+40+212+40+4×10=372 μs. The contention-free period starts with aBeacon frame 67 (FIG. 13) followed by a SIFS. With the 108 Mbit/s bitrate and with the ERP-OFDM PHY framing this requirement equals 76+10=86μs. The contention-free period ends with a CF-End frame 109 (FIG. 20).With the 108 Mbit/s bit rate and with the ERP-OFDM PHY framing thisrequirement equals to 40 μs. The remaining time within thecontention-free repetition interval is available for the contention-freedata traffic. As the granularity of the contention-free interval is one1,024 μs time unit (TU), the time available for contention-free trafficwhen the contention free interval is set to one TU is1024−290−372−86−40=236 μs. Taking into consideration the maximum dataframe size as specified by IEEE 802.11, the MAC, CCMP, and PHYencapsulation overheads and the SIFS between successive data frames,only the maximum of 17.7 Mbit/s effective user data speed can be donewith this interval set-up. With contention free interval set to seven(7) TUs, the time available for contention-free data becomes 6 352 μswhich allows the transmission of up to 32 of the 688 (or 689) sampleblocks of 24 bits each. With 32 blocks per interval each of the 16 datasources will transmit a sample data block twice. This arrangementminimizes the sampling rate and transmission rate alignment cycle andsimplifies the alignment algorithm.

To optimize the smooth flow of data and to minimize the buffering needs,the average rate of samples per TU should be kept as close to1,024/1,000×192=196.61 as possible by varying the size of the dataframes in the proper way in accordance with FIG. 23. At the same timethe data flow from each of the 16 data sources should be as smooth aspossible. The following frame size algorithm, that is one of the keyinnovations in this invention, is introduced. The contention-free timeis first split into 32 block buffers of varying size. Each buffercorresponds to an individual sequential signal source. During eachcontention-free period each of the 16 sources transmits twice making thetotal of 32 buffers. These buffers are presented as columns in FIG. 23.The buffer size varies between 688 and 689 sample records each,according to the following set of size adjustment rules. If no exceptionrule applies, the default size is 688. The exceptional blocks contain689 sample records each. The first exceptional block number x_(j1) forthe j-th data source is calculated by the formula

x _(j,1)=8 mod (13−j)+1,

resulting values 5, 4, 3, 2, 1, 8, 7, 6, 5, 4, 3, 2, 1, 8, 7, and 6 forthe signal sources from 1 to 16, respectively. After this an exceptionaldata blocks repeats after each seven default size blocks until the limitof 250 source blocks is reached. Yet another exception rule is applied.For sources 1, 2, 3, 6, 7, 8, 9, 10, 11, 14, 15, and 16 the blocks 200,221, 242, 11, 32, 53, 74, 95, 116, 137, 158, and 179 each will contain689 sample records. After 250 blocks the full cycle is repeated. Thefull cycle thus contains 125 intervals of 7 TUs each resulting a fullcycle time of 125×7×1 024 μs=896 ms. Each independent signal sourcetransmitter implements its own sequencing. This algorithm guarantees, inaccordance with FIG. 24, that the buffering jitter remains below +/−1.5sample within all the buffer sets and becomes zero at the end of each125^(th) sample buffer set. With this adjustment algorithm there is aworst-case margin of 80 μs within the contention-free data transfertime. This arrangement also makes it possible to support the effectiveuser data contention traffic of up to 5 Mbit/s along with the real-timeaudio transmission. The contention traffic is available for systemconfiguration and control as well as for other independent dataexchange.

As shown above, the choice of at least seven TUs for the duration of theBeacon Repetition interval is required to reserve enough bandwidth forthe contention-free isochronous audio traffic and to keep the ratesalignment algorithm manageable. Selecting the minimum value of seven TUsfurther minimizes the system delay and buffering requirements. Also,selecting the value of seven TUs instead of any bigger ones, creates amaximum bandwidth for the contention-based traffic, in addition to thecontention-free isochronous audio traffic.

Error Control

According to FIGS. 6 and 7, the error control method is optimised forsimplicity and speed under the assumptions of human listening ofmulti-channel studio-quality voice and music audio sound. This means asimple and fast best-effort error correction scheme that reduces theaudible effect of the errors to a non-observable level. The method takesadvantage on the long 24-bit audio data samples and the high 192kSample/s sampling rate as well as the inherent property of the extendedIEEE 802.11g implementation to transform transmission path originatedburst errors to single-bit errors in reception. However, this errorcorrection scheme is not appropriate for applications where no errorscan be tolerated.

Thanks to the WLAN transmission error correction method, almost all theresidual reception errors are single-bit errors. It is thereforesufficient to correct the effects of single bit errors. The errordetection is done by comparing a sample to the average of theimmediately preceding and following samples. If the difference is largerthan a predefined maximum inter sample difference limit then all the 24one bit variants of the sample prepared by bitwise exclusive or functionof all the bit locations are compared to the calculated average and theone with the smallest absolute difference is chosen to replace theerroneously received sample. This process is illustrated in FIG. 7.Because of the high sampling rate, the residual errors are not audibleby the human ear.

Synchronization

According to FIG. 20, the synchronization within the system is based onthe repetitive appearance of the end-of-frame interrupt generated by theCF-End frame 109 at exactly 6 802 μs after the beginning of eachrepeating 7 168 μs contention-free repetition interval. The end-of-frameinterrupt of this control message 109 synchronizes all the signaltransmitters 6 in regard of signal sampling, transmission block sizecalculation, and transmission timing within the inaccuracy of theinterrupt latency time difference of the receivers. Because all thereceivers are programmed to wait for this particular interrupt, thesystem level synchronization jitter caused by the interrupt latency isof the order of one instruction execution cycle (added with the verysmall processor-to-processor crystal oscillator phase jitter). Inpractise, this total jitter is of the order of 100 ns and cannotpossibly be noticed by human listener. For comparison, the 192 kSample/saudio sampling cycle is 5.21 μs.

Detailed Description of the WLAN Transmission Cycle

According to FIG. 25 in the idle state, when no audio signal is present,the collector receiver is programmed to run the beacon interval of onetime unit (1 TU). When the audio stream needs to be started, acontention-free mode command is sent to all transmitters using theirgroup address and the beacon interval is reprogrammed to 7 TUs of 1 024μs each totalling 7 168 μs. The CF-End end-of-frame interrupt of thisframe triggers the beginning of synchronous source signal sampling inall transmitters. The transmitters also program their hardwaretransmitter timers to be started by the same interrupt. The transmissionstart time for each signal source is determined by the timer valuegenerated by a special virtual token passing method as follows. Thepoint coordination function (PCF) is implemented in the receivercollector of the WLAN access point station. The beacon repetitioninterval, and hence the contention-free repetition interval, are set toseven time units and every such period contains a contention-free and acontention part. The length of the allocated contention-free period isset to 6 748 μs using the CFPMaxDuration parameter in the Beacon frame67 and this set-up leaves a guaranteed 290 μs for the decentralizedcontrol function (DCF) contention traffic. This time is large enough forthe transmission one maximum length data frame during the contentionperiod together with its acknowledgement and the associated inter-frameelements as required by the IEEE 802.11 standard. It also means that aminimum of 2.58 Mbit/s of bandwidth (when maximum size data frames areused) is always available for contention traffic. Under heavy traffic oflarge frames, the allocated contention-free period becomes foreshortenedfrom the beginning when a frame is being transmitted during the expectedstart of the contention-free period. Because this contention exchangecan include the CTS and ACK control frames with their associatedinter-frame elements in addition to a maximum size data frame, up to amaximum of 324 μs may be consumed by the busy medium from the beginningof the contention-free period.

The worst-case transmission-timing scenario for the audio data is asfollows. The expected beginning of the contention period occurs but amaximum length contention transfer sequence was just started. It willcause a 324 μs contention-free period foreshortening. Only after thisforeshortening delay, the 40 μs Beacon message that sets the NAVcondition, can be transmitted. The first audio data block transmissionstarts after an additional 10 μs SIFS time has elapsed. This is a totalof 374 μs after the expected beginning of the contention-free period. Inthe case of a smaller or none foreshortening, a quiet filler period isinserted by the transmitter software to reach the 374 μs tick. Thisarrangement guarantees that the first audio bit is always sent on thesame relative tick within the 6 748 μs contention-free repetitioninterval. The available transfer time for the contention-free audio datais therefore 7 168−374−290−40−10=6454 μs. In the worst-case scenario,all audio buffers contain either 688 or 688 24+8-bit sample records. Thetime needed to physically send either buffer together with their MACheaders and trailers as well as an AES based CCMP encapsulationoverheads is the same 186 μs. Each frame is followed by a 10 μs SIFSperiod. The time needed to send two full sets of sixteen blocks from thesixteen independent signal sources together with their SIFS periods isthus 2×16 (186 +10)=6 272 μs. The transmission sequence is finallyfollowed by a 80 μs programmed idle delay after which a 40 μs CF-Endbroadcast frame 109 terminates the contention-free period, alsoresetting the NAV condition initially set by the beginning of the Beaconframe. This happens exactly at the same time as the contention-freeperiod would have ended based on the timers set by the CFPMaxDurationparameter of the Beacon frame. The time margin within thecontention-free period of 80 μs out of the minimum available time of 6352 μs represents just a 1.26 percent contention-free time margin. Atthis point, the contention period starts allowing the transmission of asingle maximum size frame with an ACK response plus the associated twointer-frame SIFS times and two slot times as specified in IEEE 802.11standard.

Operation of the Transmitter and Base Station

In accordance with FIGS. 1 -3 based on the commands from the remotecontrollers 5 the system selects a recording or broadcasting subset outof the possible n AES (S/PDIF) digital outputs. The roles of the signalsources 6 are also programmed at this point with the controllers usingthe individual addresses of the signal sources 6 and their LEDindicators. Also the group address of the signal sources is set now.

In this application the speeded-up multicast means a procedure, whereall transmitters 7 transmit their data packages back-to-back using thesame group address and the end of frame interrupts triggered hardwaretimers for their transmission timing. Thus no polling and noacknowledgements are used. The first transmitter 7 is programmed totransmit 10 μs after the end of the end of frame interrupt of the Beaconframe. Other transmitters 7 are programmed to transmit 10 μs after theend of the end of frame interrupt of their predecessor's frame.Transmitter number 16 is considered the predecessor of transmitter 1.The sequence ends when each source transmitter has transmitted twice.The transmission times are listed in FIG. 25 a and illustrated in FIG.25 b. This protocol is called the simplified Virtual Token Passing(sVTP).

This invention is applicable for various isochronous data transmissionsystems, but as described here, it is particularly suitable for multichannel audio signal collection purposes.

Some video applications are also suitable for some embodiments of thepresent invention.

In addition to the WLAN transmission medium, this invention is alsoapplicable for UltraWideband radio transmission technology, or HomePlugAV type transmission technology, where the mains power cable is usedalso for data transmission. In the latter case, the transmission systemis not literally wire free, but since active loudspeakers always requireexternal power feeding through a cable, no additional cabling isrequired for data transmission.

1. An isochronous signal collection method for streaming digitalisochronous data from multiple, independent but coordinated, signalsources in a standard wireless local area network transmission systemwhere bandwidth is reserved for both contention-based traffic andcontention free traffic, the method comprising the steps of; organizingaudio data formed by samples in audio frames and sending the audioframes to a receiver using speeded-up multicasting, within consecutivebeacon intervals, and adjusting the contention free traffic of thebeacon interval to an optimum value such that, enough bandwidth isreserved for the contention-free isochronous audio traffic, the systemdelay and buffering requirements are minimized, and a maximum bandwidthfor the contention-based traffic is reserved, in addition to thecontention-free isochronous audio traffic.
 2. A method in accordancewith claim 1 further comprising the step of, setting the IEEE 802.11standard for the beacon interval to 7 time units.
 3. A method inaccordance with claim 1 further comprising the step of, varying thenumber of the samples in the audio frames in order to minimize thebuffer size in the transmitter and receivers.
 4. A method in accordancewith claim 3 further comprising the step of, varying the number ofsamples in the audio frames in a cycle of 125 consecutive beaconintervals.
 5. A method in accordance with claim 4 further comprising thestep of, varying the number of samples in the audio frames in a cycle of125 consecutive beacon intervals in accordance with a set of rules
 6. Anisochronous transmission method for collecting streaming multi channeldigital isochronous data from multiple independent sources in a standardwireless local area network transmission system, where bandwidth isreserved for both contention-based traffic and contention free traffic,the method comprising the steps of; organizing audio data formed bysamples is organized in to audio frames and sending the audio frames toa receiver using speeded-up multicasting, within consecutive beaconintervals, adjusting the contention free traffic of the beacon intervalto an optimum value, and adjusting the length of the beacon intervalsuch that a required amount of audio data can be sent to the receiverswith minimum system delay.
 7. A wireless transmission system forcollecting streaming digital serial audio data, in which systembandwidth is reserved for both contention traffic and contention freetraffic, the system comprising; an organizing means for organizing theaudio data formed by samples in audio data frames and control frames, asending means for sending the frames to the receiver within consecutivebeacon intervals, and an adjusting means for adjusting the contentionfree traffic of the beacon interval to an optimum value such that,enough bandwidth is reserved for the contention-free isochronous audiotraffic, the system delay and buffering requirements are minimized, anda maximum bandwidth for the contention-based traffic is reserved, inaddition to the contention-free isochronous audio traffic.
 8. A systemin accordance with claim 7, further comprising, a setting means forsetting in IEEE 802.11 standard the beacon interval to 7 time units. 9.A system in accordance with claim 7, further comprising; a varying meansfor varying the number of the samples in the audio frames in order tohave a smooth data flow with minimal jitter and to minimize the buffersize in transmitter and receivers.
 10. A system in accordance with claim9, wherein the varying means varies the number of samples in the audioframes in a cycle of 125 consecutive beacon intervals.
 11. A system inaccordance with claim 10, wherein the varying means varies the number ofsamples in the audio frames in a cycle of 125 consecutive beaconintervals in accordance with a set of rules.
 12. A wireless transmissionsystem for streaming digital serial audio data, in which systembandwidth is reserved to both contention traffic and contention freetraffic, the system comprising; an organizing means for organizing theaudio data formed by samples in audio data frames and control frames, asending means for sending the audio data frames and control frames to areceiver within consecutive beacon intervals, a first adjusting meansfor adjusting the contention free traffic of the beacon interval to anoptimum value, and a second adjusting means for adjusting the length ofthe beacon interval such that a required amount of audio data can besent to the receiver with minimum system delay.
 13. A method inaccordance with claim 1, the method further comprising the steps of;dividing the audio data into data blocks of a predetermined length, andwherein when a difference between an audio data sample and the averageof the preceding and following samples exceeds a predetermined limit,replacing the corresponding audio data by the nearest one bit exclusiveor function variant of the received data compared to this average. 14.(canceled)
 15. (canceled)
 16. (canceled)
 17. (canceled)
 18. A method inaccordance with claim 1, in which the method further comprises the stepsof; organizing the audio data in frames containing control frames andaudio frames, sending the organized audio data is sent by multicastingto multiple receivers within consecutive beacon intervals from multipletransmitters, and synchronizing the audio data between the multipletransmitters by an interrupt signal generated by each beacon interval.19. (canceled)
 20. A method in accordance with claim 18, wherein theinterrupt command is an end of frame interrupt command.
 21. (canceled)22. (canceled)